Resampling 22.05 KHz WAV Audio Files to 8KHz for Speech Communication Systems

This process involves resampling 22.05 KHz WAV audio files to 8KHz using MATLAB to meet the requirements of speech communication systems. Resampling is a digital signal processing technique that modifies the sampling rate of audio signals, specifically reducing the original 22.05 KHz sampling frequency to 8KHz in this implementation. The primary objective is to optimize audio files for voice communication systems, thereby enhancing speech quality and transmission efficiency. In MATLAB, this resampling operation can be efficiently performed using the resample() function, which employs anti-aliasing filters and interpolation algorithms to prevent signal distortion during sampling rate conversion. The implementation typically involves specifying the original sampling rate (22.05 KHz), target sampling rate (8KHz), and applying appropriate filter parameters to maintain audio quality while reducing the data rate. The resampling process follows these key steps: reading the original WAV file using audioread(), calculating the resampling ratio (8/22.05), applying the resample() function with optimal filter settings, and saving the output using audiowrite(). MATLAB's signal processing toolbox provides advanced algorithms including polyphase implementation and Kaiser window filtering to ensure minimal phase distortion and aliasing effects. This technical approach enables effective adaptation of audio files for bandwidth-constrained communication systems while preserving essential speech characteristics through proper downsampling techniques and digital filter design.